Mobile telephony networks have initially been conceived for allowing voice communications, in a way similar to the Public Switched Telephone Networks, shortly PSTNs, but between mobile users. The mobile telephony networks have experienced and are experiencing an enormous spread, especially after the introduction of second generation mobile network, and particularly of digital mobile networks, such as those conforming to the GSM (“Global System for Mobile communications”) standard, and to the corresponding systems adopted in the United States and in Japan.
In way similar to the PSTNs, the second generation mobile networks are circuit switched networks; this greatly limits the bandwidth that can be allocated for a given user, especially in mobile networks of the second generation. On the contrary, data communications networks such as computer networks and, among these, the Internet, adopt packet switched schemes, that allow much higher data transfer rates.
Some solutions have been proposed for overcoming the limitations of the traditional circuit switched mobile networks such as the GSM networks, so as to allow the users of mobile terminals to exploit in efficient way the services offered through the Internet. One of the solutions that is acquiring a significant popularity is the GPRS (“General Packet Radio Service”). The GPRS is a digital mobile telephony technology compatible with the GSM networks (actually, it is built on the existing GSM network architecture) that allows data transfer at a higher speed than that allowed by the pure GSM. Essentially, the GPRS can be seen as an add-on to the GSM, that supports and enables packet data communication. Although third generation wireless communications systems such as those conforming to the standard UMTS (“Universal Mobile Telecommunication System”) are more promising in terms of data transfer speed, the GPRS is a within reach solution to improve the data exchange capability in existing GSM networks.
The services offered by these mobile networks in addition to the simple vocal communications have quickly increased in number and quality; just to cite some examples, in the last few years short messaging services (“Short Messaging System”, shortly SMS) and multimedia messaging services (“Multimedia Messaging System”, or MMS), and Internet connectivity services have been made available.
In particular, there is a strong interest in providing multimedia services to the users of mobile communications networks, i.e., services thanks to which there is the possibility of adding images, video, access to data through the Internet or through the electronic mail, to a communication between users that is made of voice alone. Among these services, the so-called “combinational services” are attracting great attention of the mobile telephony operators. For the purposes of the present description, by “combinational service” there is, in general, intended a service through which a terminal of a (not necessarily mobile) communications network can simultaneously open and use two connections, typically a circuit (circuit-switched or CS) connection and a packet (packet-switched, PS) connection.
U. Olsson and M. Nilsson, in the article “Combinational services—The pragmatic first step toward all-IP”, Ericsson Review No. 2, 2003, describe, inter alia, an example of so-called “combinational services”, in which the ability to simultaneously handle traffic on a circuit connection and on a packet connection is used: the sharing of images during a conversation. The authors notice that the possibility of simultaneously handle traffic on a circuit connection and on a packet connection is allowed both with the WCDMA (Wideband Code Division Multiple Access), giving the possibility to use multiple and parallel bearers in the “over-the-air” interface (multiple Radio Access Bearers, multi-RAB), and with the GSM, in which a standardized mechanism—the Dual Transfer Mode, or DTM—yields similar possibilities. In the article, the authors notice however that the mere technical possibility of “successfully crossing the air” is not enough. Sometimes it is forgotten that the average final user is not interested in the complications of the channels coding and wave propagation. Instead, the final user wants a mobile terminal that is reliable, simple to use, and well adapted to the current context. In other words, some entity in the mobile terminal has to interpret what the user is trying to do and translate it into a sequence of operations. An example described in the article relates to a woman and her husband during a conversation about an extraordinary gardens exposition. During the conversation, the woman decides to show to her husband what she is describing. Ideally, the user interface (Man-Machine interface or MMI) should be sufficiently simple, in such a way that she only needs to press the key for turning the video camera on. The mobile terminal should contain enough intelligence to discover how to reach the other participant to the call on a packet connection and send the images. According to the authors, in order to let this happen, the following base blocks are necessary:                a coordination function in the mobile terminal (at least, an extended address book with information related to the reachability of all the pertinent networks);        a reachability mechanism in the packet connection part. According to the authors, the long term solution will probably be based on IMS (IP Multimedia Subsystem), using a SIP (Session initiation Protocol) session protocol to find the other participant and negotiate the session parameters;        a mechanism for distributing informative capability that allows to applications based on the terminals and on the network to make an intelligent use of the information related to the subscription, the session state, the bearer states, user preferences and so on. In this respect, the authors suggest, among the other things, the SIP, the use of the HTTP protocol and the Web XML services.        
Another description of a service enjoyable through mobile terminals is given in the White Paper “Video sharing—Enrich your voice call with video”, by Nokia Corporation, publicly available for download at the Internet site:                http://www.nokia.com/BaseProject/Sites/NOKIA_MAIN_18022/CDA/Categories/Phones/Technologies/VideoSharing/_Content/_Static_Files/video_sharing_a4_2510.pdf        
The real time video sharing service allows the users, during a telephone conversation, to easily enrich their communication. One or the other of the users can share a live video taken by a video camera or video clips from the terminal. Both the users see the same video and can discuss about it while they are continuing their voice call. In an example described in the White Paper, Keith and Malcom are in a normal call on a circuit connection (CS) and Keith wants to share live video with Malcom. They both have devices capable of video sharing and are registered for the service. The following flow is described in the White Paper:                During the ongoing CS voice call, Keith chooses to share the live video.        Keith confirms Malcom as a recipient.        Malcom receives video request from Keith and accepts it.        The system shows the acceptance to Keith, who activates the sending of the video stream.        Malcom's device starts showing the same video as Keith's device. They can discuss it via the voice call.        Keith ends the video sharing when he has shown what he wanted. The voice call between Keith and Malcom remains active.        
The technology used by the video sharing service described by the above-mentioned White Paper is the general SIP or IMS infrastructure.
In another White Paper, entitled “White Paper IP Convergence Based On SIP: Enhanced Person-To-Person Communications”, publicly made available by Forum Nokia for download at the Internet site                http://www.forum.nokia.com/main/1,040,00. html?fsrParam=2-3-/main.html&flleID=5336the use of the SIP protocol is described for the establishment of peer-to-peer connections type between terminals capable of supporting the IP protocol. As described in the White Paper, in order to communicate, IP-based applications must have a mechanism to reach the correspondent. Today, fixed and mobile telephony networks perform this critical task of establishing a connection. By dialing the other user's telephone number, the network can establish an ad-hoc connection between any two terminals. This critical connectivity capability still does not exist widely in the Internet. According to this White Paper, SIP-based sessions management, complemented by other critical mobile networks capabilities (i.e., authentication, roaming, and network interconnection provided by the IMS standard) provides the required structure. With the implementation of such a system, it is possible to establish an IP connection between two terminals. Once the connection is established, it can be used to exchange all types of communication media (voice, video, content, etc.). Like HTTP, SIP is a text-based client-server protocol. SIP was designed to establish, modify and terminate multimedia sessions or calls, and it differs from the HTTP in the fact that a “SIP terminal” (or User Agent, UA) may act as both a client and a server. Therefore it is possible to establish a client-to-client communication. The version 6 of the IP protocol (IPv6) provides, according to this White Paper, a robust base for the SIP-based services. The most important benefit of IPv6 is the large address space. In fact, a problem also underlined in this White Paper is that the version 4 of the protocol IP (IPv4) and that such protocol has been designed for a smaller number of Internet hosts, compared to what the Internet is experimenting today. The problem of the address space will become even greater when hundreds of million (or billions) of cellular handsets will be connected to the Internet. Among the possible applications of the SIP protocol described in this White Paper, the real time video sharing service is cited.        